what about exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
regards 2015-04-08 5:45 GMT+00:00 Dmitriy Serov <serov....@gmail.com>: > Hi, Andrew. > > You are trying to solve two tasks: definition through what line the call > came and a beautiful display of this information. > 1. definition through what line the call came. If the username and > password for inbound and outbound registration the same, then try the > following: > a) delete "register" lines. > b) add option "callbackextension=Company1" to Company1 friend section.. > And in others with their names too. > or you can change "/s" to "/Company1" in register line. > > 2. beautiful display of this information > a) add option "setvar=fromCompany=Company1" to Company1 friend section.. > b) In dialplan add > Set(CALLERID(name)=${fromCompany} ${CALLERID(name)}) > > Maybe this will help? > > Dmitiy. > > 08.04.2015 2:48, Andrew Galdes пишет: > > Hi Dmitriy and others and thanks for your help so far. > > The option "match_auth_username=yes" seems to have had no effect. From > my reading, this option will try to match the username of the incoming SIP > account to a section heading. If that is how it must work then i can see a > big problem. I'm trying to present the receptionist with a nice display of > which line the call came in on. For example, the receptionist answers calls > for 8 different companies and would like the phone to display the company > name that she should announce to the caller. > > Here is a more complete output of an incoming call. I've changed the SIP > numbers to "Company1', etc, to hide the numbers. > > Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267) >> Verbosity is at least 12 >> asterisk*CLI> >> asterisk*CLI> >> asterisk*CLI> >> == Using SIP RTP CoS mark 5 >> -- Executing [s@incoming:1] *Set*("*SIP/Company1-00000797*", >> "*thedid=""NodePhone"<sip:compa...@sip.internode.on.net >> <sip%3acompa...@sip.internode.on.net>>"*") in new stack >> -- Executing [s@incoming:2] *Set*("*SIP/**Company1**-00000797*", " >> *pseudodid="NodePhone"<sip:** sip:Company2**@sip.internode.on.net >> <http://sip.internode.on.net>>*") in new stack >> -- Executing [s@incoming:3] *Set*("*SIP/**Company1**-00000797*", " >> *pseudodid="NodePhone"<sip:** sip:Company2*") in new stack >> -- Executing [s@incoming:4] *Set*("*SIP/**Company1**-00000797*", " >> *pseudodid=** sip:Company2*") in new stack >> -- Executing [s@incoming:5] *GotoIf*("*SIP/**Company1**-00000797*", " >> *0?internal,33,1:6*") in new stack >> -- Goto (incoming,s,6) >> -- Executing [s@incoming:6] *GotoIf*("*SIP/**Company1**-00000797*", " >> *0?internal,88,1:7*") in new stack >> -- Goto (incoming,s,7) >> -- Executing [s@incoming:7] *GotoIf*("*SIP/**Company1**-00000797*", " >> *0?internal,36,1:8*") in new stack >> -- Goto (incoming,s,8) >> -- Executing [s@incoming:8] *GotoIf*("*SIP/**Company1**-00000797*", " >> *1?internal,36,1:9*") in new stack >> -- Goto (internal,36,1) >> -- Executing [36@internal:1] *Set*("*SIP/**Company1**-00000797*", " >> *CALLERID(name)=SIP/**Company1**-00000797*") in new stack >> -- Executing [36@internal:2] *Dial*("*SIP/**Company1**-00000797*", " >> *SIP/36,20*") in new stack >> == Using SIP RTP CoS mark 5 >> -- Called SIP/36 >> -- SIP/36-00000798 is ringing >> == Spawn extension (internal, 36, 2) exited non-zero on >> 'SIP/Company1-00000797' >> asterisk*CLI> exit > > > And here is the "sip.conf": > > [general] >> match_auth_username=yes >> register=081...:...@sip.internode.on.net/s >> register=082...:...@sip.internode.on.net/s >> register=083...:...@sip.internode.on.net:/s >> register=084...:...@sip.internode.on.net:/s >> register=085...:...@sip.internode.on.net/s >> register=086...:...@sip.internode.on.net/s >> register=087...:...@sip.internode.on.net/s >> register=088...:...@sip.internode.on.net/s >> >> [Company1] >> username=081... >> fromuser=081... >> secret=... >> canreinvite=no >> qualify=yes >> context=incoming >> type=friend >> insecure=invite,port >> fromdomain=sip.internode.on.net >> host=sip.internode.on.net >> dtmfmode=rfc2833 >> disallow=all >> allow=alaw >> allow=ulaw >> allow=g729 >> bindport=5060 >> bindaddr=0.0.0.0 >> nat=yes >> registertimeout=5 >> allowoverlap=no >> srvlookup=no >> ubscribecontext=from-sip >> callcounter=yes > > > > [Company2] >> ... >> [Company3] >> ... >> [Company4] >> ... > > And here is some of the "extensions.conf" file: > > [incoming] >> ; Get the DID number from the TO header. >> exten => s,1,Set(thedid="${SIP_HEADER(TO)}") >> exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) >> exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) >> exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)}) > > >> ; Direct the DID accordingly. >> exten => s,5,GotoIf($["${pseudodid}" = "081"]?internal,33,1:6) >> exten => s,6,GotoIf($["${pseudodid}" = "082"]?internal,88,1:7) >> exten => s,7,GotoIf($["${pseudodid}" = "083"]?internal,36,1:8) >> exten => s,8,GotoIf($["${pseudodid}" = "084"]?internal,36,1:9) >> exten => s,9,GotoIf($["${pseudodid}" = "085"]?internal,36,1:10) >> exten => s,10,GotoIf($["${pseudodid}" = "086"]?internal,89,1:11) >> exten => s,11,GotoIf($["${pseudodid}" = "087"]?internal,36,1:12) >> exten => s,12,GotoIf($["${pseudodid}" = "088"]?internal,13,1:13) > > > > -Andrew Galdes > > > On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov <serov....@gmail.com> wrote: > >> >> This is one of the chronic problems. Try this option in sip.conf: >> match_auth_username=yes >> >> Carefully read the description, it is better to test in "after hours". >> >> 02.04.2015 2:50, Andrew Galdes пишет: >> >> Hello all, >> >> I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts >> with the same service provides. We have 8 phone numbers in total. >> >> Incoming calls from the public are all correctly directed to >> appropriate office handsets. However, the display on the reception phone >> (the only one i care about) is always showing the same >> "SIP/Account1_0843214321" rather than the account representing the number >> dialed. >> >> For-instance, if Sam on her mobile calls "*0811111111*", Asterisk will >> show a log entry like the following: >> >> -- Executing [s@incoming:1] Set("SIP/*Account1_0822222222*", " >> thedid=""NodePhone"<sip:*0811111111*@sip.internode.on.net>"") in new >> stack >> But "Account1_*0822222222*" (as the name suggests) has a phone number >> of "*0822222222*" and not "*0811111111*". >> >> So Sam's call will come through and be routed to the correct handset as >> the business needs, but it seems that all incoming calls are being labeled >> as though coming in on a different account. The effective problem is that >> the calledID is now wrong. >> >> I'm after some general advice on how to handle the problem. >> >> Ta, >> >> >> -Andrew >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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